This invention relates to a method and apparatus for connecting a communication device such as a telephone via an IP (Internet Protocol) network. More particularly, the invention relates to a communication device connecting method and apparatus in which when a call originate request has been issued to make a connection to a called terminal via an IP network, the originating terminal is connected to the terminating terminal upon taking into account the communication quality of the route through the IP network.
IP telephone systems for placing telephone calls via an IP network currently have been spotlighted as telephone systems for the next generation to supplant existing telephone systems based upon STM (Synchronous Transfer Mode) networks. Arranging it so that ease of use and convenience on a par with that of existing telephone schemes can be realized is an important challenge confronting IP telephony. Various modes of utilizing conventional IP telephone systems have been proposed and implemented. One such example is an Internet relay telephone service provided to general users such as individuals by an Internet telephone service provider. The purpose of this system is to lower the cost of calls by using the Internet as an intermediary.
FIG. 9A is a diagram useful in describing such an Internet relay telephone service. When the user of a telephone 1 dials up an access point of an Internet telephone service provider, an IP packetizing unit 3 responds via an exchange (STM telephone network) 2 and allows the user to enter the telephone number of the call destination. On the basis of the entered telephone number, the IP packetizing unit 3 queries a server 5, referred to as a “gatekeeper”, via an IP network 4 with regard to the IP address of an IP packetizing unit 6 at the destination of transferred packets. Upon learning of the IP address, the IP packetizing unit 3 executes a procedure (stipulated by Recommendation H.225.0) for setting up an IP connection to the IP packetizing unit 6 of the called party. If the IP connection is established, then the telephone call is implemented via a route constituted by the originating telephone 1, the exchange 2, the IP packetizing unit 3, the IP network 4, the IP packetizing unit 6, an exchange 7 and a terminating telephone 8 in the order mentioned. It should be noted, however, that the transfer of actual voice data (voice packets) is performed between the IP packetizing units 3 and 6 using RTP (Real Time Protocol).
Thus, to place a telephone call from an originating telephone to a terminating telephone via an IP network in accordance with the prior art, it will suffice to dial the IP-network access telephone number and the telephone number of the terminating telephone. To make a telephone call via an existing STM network, only the telephone number of the terminating telephone need be dialed.
An IP network offers best-effort traffic service and uses the network band to the fullest extent possible; it does not compensate for bandwidth and communication quality. As a consequence, if the IP network 4 becomes congested, packet loss PL occurs, as shown in FIG. 9B, a packet delay PD is produced and delay time varies owing to the characteristic of the IP network. Thus, though the IP network offers the advantage of low communication cost, there are instances where voice is delayed or interrupted by congestion, resulting is degraded voice quality. This differs from the case where use is made of an STM network.
The state of the art is such that if an IP network has been accessed to make a telephone call, the call must be continued regardless of whether voice quality is good or bad. If voice quality is poor and the caller wishes to re-connect via an STM network, the user must take the trouble to first break the connection and then redial.